Summary: | ASTERISK-08477: Transfering of calls does not work in 1.4 through chan_agent | ||
Reporter: | Peter Zieba (pzieba) | Labels: | |
Date Opened: | 2007-01-03 22:19:02.000-0600 | Date Closed: | 2007-02-16 19:22:20.000-0600 |
Priority: | Blocker | Regression? | No |
Status: | Closed/Complete | Components: | Core/General |
Versions: | Frequency of Occurrence | ||
Related Issues: | |||
Environment: | Attachments: | ||
Description: | When trying to transfer a call, only one side of audio is present after tranfer is complete. Using phones "A" "B" and "C", the situation is illustrated as follows: "A" calls "B" "A" & "B" can hear eachother "B" attempts to tranfer to "C" "B" & "C" can hear eachother. "A" listens to on-hold music. "B" Completes transfer "A" can hear "C". "C" cannot hear "A" Blind tranfers seem to work properly. If a call that comes in on a queue (AgentCallBackLogin is being used) is transfered, Asterisk goes into an inconsistent state, while printing messages such as: [Jan 3 21:27:50] WARNING[14020] chan_zap.c: Ring requested on channel 0/1 already in use on span 1. Hanging up owner. No further calls will be taken from the PRI past this point until Asterisk is restarted. If a call that comes in on a queue is blind-tranfered, things seem to work properly. ****** ADDITIONAL INFORMATION ****** CentOS 4.4 (Final) 09:04.0 Communication controller: Digium, Inc. Wildcard TE410P Quad-Span togglable E1/T1/J1 card 3.3v (rev 02) Dell Poweredge 2850 2.8GHz 2GB RAM Polycom IP430 Phones | ||
Comments: | By: Peter Zieba (pzieba) 2007-01-03 22:26:42.000-0600 All phones are on the same network. Believe problem occured in 1.4 (compiled tarball). As far as I can remember, problem was not present in 1.2.10 (although there were problems with call queues in 1.2.10 using agentcallback login, tranfer functionality unreleated to queues worked properly) By: Peter Zieba (pzieba) 2007-01-03 22:28:06.000-0600 Calls coming in from queue are coming from the TE410P. Example illustrating phones A, B, and C only includes Sip phones on a dedicated VoIP lan. By: Peter Zieba (pzieba) 2007-01-03 22:50:27.000-0600 Ignoring the queue aspect of the problem, this seems to be a duplicate of issue 0008696 By: Serge Vecher (serge-v) 2007-01-04 08:20:53.000-0600 alright, since you are using sip devices, let's verify what the problem is by producing the log as outlined in note 0057086 in 8696 By: Peter Zieba (pzieba) 2007-01-04 16:08:29.000-0600 I can try testing some of this later tonight while there are no call coming in so I can provide you guys with some logs, but I cannot simulate this setup as I do not have an extra PRI to play with. I will be rolling back to Asterisk 1.2.14 at midnight. By: Joshua C. Colp (jcolp) 2007-01-25 21:30:44.000-0600 Any update on this? Logs/whether it is solved/etc? By: Joshua C. Colp (jcolp) 2007-02-16 19:22:20.000-0600 Fixed in 1.2 as of revision 55073, 1.4 as of revision 55086, and trunk as of revision 55087. Peace! |